Summary: | ASTERISK-24557: WebRTC call returns error "Failed to get local SDP" | ||
Reporter: | Osaulenko Alexander (a.osaulenko) | Labels: | |
Date Opened: | 2014-11-25 07:47:31.000-0600 | Date Closed: | 2014-12-23 13:58:30.000-0600 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | |
Versions: | 11.9.0 11.14.1 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | centos6.6, debian7.7.0 (i686 and x64) | Attachments: | ( 0) backtrace_20141222.txt ( 1) backtrace.txt ( 2) backtraceSDP_fail.txt |
Description: | We use Asterisk of different versions 11.9-11.14.1 with default settings for WebRTC and Openssl version: OpenSSL 1.0.1e 11 Feb 2013. It works but craches on:
d1_both.c(278): OpenSSL internal error, assertion failed: s->init_num == (int)s->d1->w_msg_hdr.msg_len + DTLS1_HM_HEADER_LENGTH We tried update openssl to OpenSSL 1.0.1j and then compiled Asterisk. When we try to call using WebRTC we have error "Failed to get local SDP" | ||
Comments: | By: Osaulenko Alexander (a.osaulenko) 2014-11-25 07:55:20.794-0600 Attached backtrace.txt after Asterisk had crashed By: Matt Jordan (mjordan) 2014-11-25 09:08:12.126-0600 Thank you for your bug report. In order to move your issue forward, we require a backtrace[1] from the core file produced after the crash. Also, be sure you have DONT_OPTIMIZE enabled in menuselect within the Compiler Flags section, then: make install After enabling, reproduce the crash, and then execute the backtrace[1] instructions. When complete, attach that file to this issue report. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace What you've attached is not a valid backtrace. Please following the linked instructions. By: Osaulenko Alexander (a.osaulenko) 2014-11-27 08:25:08.449-0600 Matt, backtrace is attached By: Badalian Vyacheslav (slavon) 2014-11-28 21:22:00.667-0600 https://rt.openssl.org/Ticket/Display.html?id=3592 guest / guest By: Rusty Newton (rnewton) 2014-12-15 09:43:03.723-0600 Your trace is incomplete for what we need. Also, there is a lot of this in your trace/log: {noformat} Nov 26 07:03:51] WARNING[4696]: loader.c:452 load_dynamic_module: Module 'res_ari' did not register itself during load [Nov 26 07:03:51] WARNING[4696]: loader.c:918 load_resource: Module 'res_ari' could not be loaded. [Nov 26 07:03:51] WARNING[4696]: loader.c:452 load_dynamic_module: Module 'res_stasis' did not register itself during load [Nov 26 07:03:51] WARNING[4696]: loader.c:918 load_resource: Module 'res_stasis' could not be loaded. [Nov 26 07:03:51] WARNING[4696]: loader.c:840 inspect_module: Module 'res_ari_playbacks.so' was not compiled with the same compile-time options as this version of Asterisk. [Nov 26 07:03:51] WARNING[4696]: loader.c:841 inspect_module: Module 'res_ari_playbacks.so' will not be initialized as it may cause instability. [Nov 26 07:03:51] WARNING[4696]: loader.c:931 load_resource: Module 'res_ari_playbacks.so' could not be loaded. {noformat} Please recompile Asterisk completely (including all modules). When doing so, make sure to [follow the linked instructions|https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationAfterACrash] to use appropriate compiler flags and get a backtrace from the core after it dumps. When recompiling Asterisk, please test with the latest of the SVN 11 branch as there are a few recent fixes that may apply to your issue. By: Osaulenko Alexander (a.osaulenko) 2014-12-23 01:33:25.155-0600 Hello, Rusty Newton! I have attached new backtrace. We had made two calls and get error "Failed to get local SDP". First call was general, second was with sip debug. By: Osaulenko Alexander (a.osaulenko) 2014-12-23 03:24:46.804-0600 Hello, Rusty Newton! I have attached new backtrace - backtrace(SDP fail).txt By: Matt Jordan (mjordan) 2014-12-23 13:58:25.187-0600 Looking at your log file, the error message is being returned from whatever client Asterisk is communicating with: {noformat} <--- SIP read from WS:192.168.34.145:2562 ---> SIP/2.0 603 Failed to get local SDP Via: SIP/2.0/WS 192.168.34.199:5060;rport=5060;branch=z9hG4bK589fe139 From: "156"<sip:156@192.168.34.199>;tag=as451e8620 To: <sip:157@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=8W672ie5gHvCPbzyA7VQ Call-ID: 597126967b118c560a03e3eb6393f8da@192.168.34.199:5060 CSeq: 102 INVITE Content-Length: 0 Reason: SIP; cause=603; text="Failed to get local SDP" <-------------> {noformat} You'll need to investigate it to determine what it doesn't like about the INVITE request Asterisk sent to it. {noformat} Reliably Transmitting (NAT) to 192.168.34.145:2562: INVITE sip:157@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.34.199:5060;branch=z9hG4bK589fe139;rport Max-Forwards: 70 From: "156" <sip:156@192.168.34.199>;tag=as451e8620 To: <sip:157@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws> Contact: <sip:156@192.168.34.199:5060;transport=WS> Call-ID: 597126967b118c560a03e3eb6393f8da@192.168.34.199:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-11-r429539 Date: Mon, 22 Dec 2014 07:40:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 409 v=0 o=root 1170437591 1170437591 IN IP4 192.168.34.199 s=Asterisk PBX SVN-branch-11-r429539 c=IN IP4 192.168.34.199 t=0 0 m=audio 12760 UDP/TLS/RTP/SAVPF 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=connection:new a=setup:actpass a=fingerprint:SHA-256 62:8E:8B:50:E7:8D:D6:62:DE:3B:9E:D8:E5:B8:49:04:23:E8:34:AF:C5:93:94:FF:DD:4E:9A:32:8D:C7:B8:82 a=sendrecv {noformat} That being said, it appears as if you've modified Asterisk. Asterisk does not include the request line modifier "rtcweb-breaker" anywhere. {{INVITE sip:157@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0}} The project does not provide source for modified versions of Asterisk. |